Digital Radio Mondiale (DRM) is an international standard (used in India, Russia, South Africa, Brazil, etc.) for radio broadcasts at frequencies below 30 MHz as described in ‘DRM: System Specification, ETSI ES 201 980 V4.1.1 (2014-01)’. In standard DRM systems, there are three channels, a Main Service Channel (MSC) that carries the multiplexed audio and data signals, a Service Description Channel (SDC) that carries information needed to decode the MSC, and a Fast Access Channel (FAC) that carries OFDM (orthogonal frequency division multiplexing) properties and configuration information for the SDC/MSC. As specified by the DRM standards, the FAC is always 4-QAM and therefore more immune to fading. Typically, DRM has four transmission modes; A, B, C and D. Mode A is used on medium and long wave frequencies using ground wave propagation for local broadcasts. Mode B is used for single-hop, short wave broadcasts and is commonly used in Europe. Mode C is used for long path multi-hop broadcasts on short wave, and mode D is used for Near Vertical Incidence Skywave (NVIS) broadcasts.
A standard radio receiver system has an antenna connected to a tuner for the reception of radio signals, and the output from the radio receiver is an audio signal and/or displayed information (text or picture). For known systems the received signals may be Amplitude Modulation (AM), Frequency Modulation (FM), or Digital signals. The received signals will be decoded by a combination of hardware circuits, hardware blocks and associated software modules for playing the audio and/or displaying the text information. A baseband processor (typically a Digital signal Processor) will contain software and specialized hardware to realize the baseband algorithms used in decoding the radio signal, and extracting the audio and/or text/picture information. A control processor (typically a host processor) will exercise all the control functionalities by performing the overall coordination for playing a radio station, such as:                Sending tune commands to the tuner, and validating the tuner reply to confirm tuning is operational.        Requesting the baseband processor to send periodic notifications on received signal quality, and associated parameters.        Forwarding the decoded radio signal to the control processor by the baseband processor for audio decoding.        Streaming the final decoded signal to a speaker or other output.        
The memory and hardware circuits of the receiver work in conjunction with the processors or tuner for achieving the software realization of the modules and/or performing the functionality as specified by the hardware block.
Time to audio (TTA) is a well-known standard metric that is used to benchmark radio performance and is defined as the duration of time taken from the point at which the radio receives a radio signal/data to the time at which an audible signal is played at the speaker. Standards in this area will specify a minimum acceptable TTA.
Time-To-Audio (TTA) is typically comprised of the following parameters:                1. Tuner delay: This is well-known and generally does not need to be considered.        2. Inner receiver signal processing including signal detection, estimate of DRM transmitted signal compensated for propagation channel effects.        3. Outer receiver signal processing: channel decoding to retrieve the encoded audio data.        4. Audio decoder: Decoding the encoded audio data        5. Audio rendering delay: The delay in rendering the digital samples to the loudspeaker output. This is also a well-known and need not be accounted for.        
FIG. 1 shows a prior art channel estimator 100 for a radio receiver that uses Wiener channel estimation in the signal processing. As shown, signal 101 is received at the channel estimator 100 from a fast Fourier Transform unit (not shown). Channel estimator 100 includes a two-dimensional (2D) Wiener filter that is split into two independent components, a time direction one-dimensional (1D) filter 102, and a frequency direction one-dimensional (1D) filter 104. Time and frequency direction filter coefficients are calculated by Wiener estimation at 106 and 108 respectively.
The channel estimation is based on interpolation by fixed length adaptive Wiener-filtering in the time and frequency directions. Calculation of Wiener filters for both directions (time and frequency) uses a Minimum Mean Square Error (MMSE) solution:
      h    =                  R        hp            ·              R        pp                  -          1                    ·      p                  R      pp        =                  R        pp            +                        1          SNR                ⁢        I            Where:                h is the optimal filter coefficients,        p—is the gain reference estimates as extracted from the received signal,        Rhp—is the cross-correlation matrix between h and p,        Rpp—is the auto-correlation matrix of p,        I—is the identity matrix of Rpp size.        SNR—is estimated signal to noise ratio.        
Rhp and Rpp are independently estimated for the time and frequency domains, and an auto-regressive fit (using a levinson durbin algorithm) is then performed to estimate the channel estimates both in the time domain 106 and then in frequency domain 108. Interpolation is done firstly in a time direction and secondly in a frequency direction to arrive at the channel estimates for a particular channel.
A two-dimensional Wiener channel estimator is an optimal solution, but this is complex. Here the filter length(s) of a time direction 1D filter and a frequency direction 1D filter are decided based on the transmission mode (A-D). The particular filer length of the Wiener filter metrics introduces a considerable delay in the signal processing. At present, the ETSI standard specifies 3.2 seconds for the TTA delay, of which the Wiener filter delay is a significant part.
It is desirable to have a radio receiver where the TTA can be lowered to meet the ETSI standard.